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stream.go
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stream.go
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// Package gostream implements a simple server for serving video streams over WebRTC.
package gostream
import (
"context"
"errors"
"image"
"sync"
"time"
"github.com/edaniels/golog"
"github.com/google/uuid"
// register screen drivers.
_ "github.com/pion/mediadevices/pkg/driver/microphone"
"github.com/pion/mediadevices/pkg/prop"
"github.com/pion/mediadevices/pkg/wave"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
"go.viam.com/utils"
"go.viam.com/rdk/gostream/codec"
"go.viam.com/rdk/rimage"
utils2 "go.viam.com/rdk/utils"
)
// A Stream is sink that accepts any image frames for the purpose
// of displaying in a WebRTC video track.
type Stream interface {
internalStream
Name() string
// Start starts processing frames.
Start()
WriteRTP(pkt *rtp.Packet) error
// Ready signals that there is at least one client connected and that
// streams are ready for input. The returned context should be used for
// signaling that streaming is no longer ready.
StreamingReady() (<-chan struct{}, context.Context)
InputVideoFrames(props prop.Video) (chan<- MediaReleasePair[image.Image], error)
InputAudioChunks(props prop.Audio) (chan<- MediaReleasePair[wave.Audio], error)
// Stop stops further processing of frames.
Stop()
}
type internalStream interface {
VideoTrackLocal() (webrtc.TrackLocal, bool)
AudioTrackLocal() (webrtc.TrackLocal, bool)
}
// MediaReleasePair associates a media with a corresponding
// function to release its resources once the receiver of a
// pair is finished with the media.
type MediaReleasePair[T any] struct {
Media T
Release func()
}
// NewStream returns a newly configured stream that can begin to handle
// new connections.
func NewStream(config StreamConfig) (Stream, error) {
logger := config.Logger
if logger == nil {
logger = golog.Global()
}
if config.VideoEncoderFactory == nil && config.AudioEncoderFactory == nil {
return nil, errors.New("at least one audio or video encoder factory must be set")
}
if config.TargetFrameRate == 0 {
config.TargetFrameRate = codec.DefaultKeyFrameInterval
}
name := config.Name
if name == "" {
name = uuid.NewString()
}
var trackLocal *trackLocalStaticSample
if config.VideoEncoderFactory != nil {
trackLocal = newVideoTrackLocalStaticSample(
webrtc.RTPCodecCapability{MimeType: config.VideoEncoderFactory.MIMEType()},
"video",
name,
)
}
var audioTrackLocal *trackLocalStaticSample
if config.AudioEncoderFactory != nil {
audioTrackLocal = newAudioTrackLocalStaticSample(
webrtc.RTPCodecCapability{MimeType: config.AudioEncoderFactory.MIMEType()},
"audio",
name,
)
}
ctx, cancelFunc := context.WithCancel(context.Background())
bs := &basicStream{
name: name,
config: config,
streamingReadyCh: make(chan struct{}),
videoTrackLocal: trackLocal,
inputImageChan: make(chan MediaReleasePair[image.Image]),
outputVideoChan: make(chan []byte),
audioTrackLocal: audioTrackLocal,
inputAudioChan: make(chan MediaReleasePair[wave.Audio]),
outputAudioChan: make(chan []byte),
logger: logger,
shutdownCtx: ctx,
shutdownCtxCancel: cancelFunc,
}
return bs, nil
}
type basicStream struct {
mu sync.RWMutex
name string
config StreamConfig
started bool
streamingReadyCh chan struct{}
videoTrackLocal *trackLocalStaticSample
inputImageChan chan MediaReleasePair[image.Image]
outputVideoChan chan []byte
videoEncoder codec.VideoEncoder
audioTrackLocal *trackLocalStaticSample
inputAudioChan chan MediaReleasePair[wave.Audio]
outputAudioChan chan []byte
audioEncoder codec.AudioEncoder
// audioLatency specifies how long in between audio samples. This must be guaranteed
// by all streamed audio.
audioLatency time.Duration
audioLatencySet bool
shutdownCtx context.Context
shutdownCtxCancel func()
activeBackgroundWorkers sync.WaitGroup
logger golog.Logger
}
func (bs *basicStream) Name() string {
return bs.name
}
func (bs *basicStream) Start() {
bs.mu.Lock()
defer bs.mu.Unlock()
if bs.started {
return
}
bs.started = true
close(bs.streamingReadyCh)
bs.activeBackgroundWorkers.Add(4)
utils.ManagedGo(bs.processInputFrames, bs.activeBackgroundWorkers.Done)
utils.ManagedGo(bs.processOutputFrames, bs.activeBackgroundWorkers.Done)
utils.ManagedGo(bs.processInputAudioChunks, bs.activeBackgroundWorkers.Done)
utils.ManagedGo(bs.processOutputAudioChunks, bs.activeBackgroundWorkers.Done)
}
func (bs *basicStream) WriteRTP(pkt *rtp.Packet) error {
return bs.videoTrackLocal.rtpTrack.WriteRTP(pkt)
}
func (bs *basicStream) Stop() {
bs.mu.Lock()
defer bs.mu.Unlock()
if !bs.started {
close(bs.streamingReadyCh)
}
bs.started = false
bs.shutdownCtxCancel()
bs.activeBackgroundWorkers.Wait()
if bs.audioEncoder != nil {
bs.audioEncoder.Close()
}
if bs.videoEncoder != nil {
if err := bs.videoEncoder.Close(); err != nil {
bs.logger.Error(err)
}
}
// reset
bs.outputVideoChan = make(chan []byte)
bs.outputAudioChan = make(chan []byte)
ctx, cancelFunc := context.WithCancel(context.Background())
bs.shutdownCtx = ctx
bs.shutdownCtxCancel = cancelFunc
bs.streamingReadyCh = make(chan struct{})
}
func (bs *basicStream) StreamingReady() (<-chan struct{}, context.Context) {
bs.mu.RLock()
defer bs.mu.RUnlock()
return bs.streamingReadyCh, bs.shutdownCtx
}
func (bs *basicStream) InputVideoFrames(props prop.Video) (chan<- MediaReleasePair[image.Image], error) {
if bs.config.VideoEncoderFactory == nil {
return nil, errors.New("no video in stream")
}
return bs.inputImageChan, nil
}
func (bs *basicStream) InputAudioChunks(props prop.Audio) (chan<- MediaReleasePair[wave.Audio], error) {
if bs.config.AudioEncoderFactory == nil {
return nil, errors.New("no audio in stream")
}
bs.mu.Lock()
if bs.audioLatencySet && bs.audioLatency != props.Latency {
return nil, errors.New("cannot stream audio source with different latencies")
}
bs.audioLatencySet = true
bs.audioLatency = props.Latency
bs.mu.Unlock()
return bs.inputAudioChan, nil
}
func (bs *basicStream) VideoTrackLocal() (webrtc.TrackLocal, bool) {
return bs.videoTrackLocal, bs.videoTrackLocal != nil
}
func (bs *basicStream) AudioTrackLocal() (webrtc.TrackLocal, bool) {
return bs.audioTrackLocal, bs.audioTrackLocal != nil
}
func (bs *basicStream) processInputFrames() {
frameLimiterDur := time.Second / time.Duration(bs.config.TargetFrameRate)
defer close(bs.outputVideoChan)
var dx, dy int
ticker := time.NewTicker(frameLimiterDur)
defer ticker.Stop()
for {
select {
case <-bs.shutdownCtx.Done():
return
default:
}
select {
case <-bs.shutdownCtx.Done():
return
case <-ticker.C:
}
var framePair MediaReleasePair[image.Image]
select {
case framePair = <-bs.inputImageChan:
case <-bs.shutdownCtx.Done():
return
}
if framePair.Media == nil {
continue
}
var initErr bool
func() {
if framePair.Release != nil {
defer framePair.Release()
}
var encodedFrame []byte
if frame, ok := framePair.Media.(*rimage.LazyEncodedImage); ok && frame.MIMEType() == utils2.MimeTypeH264 {
encodedFrame = frame.RawData() // nothing to do; already encoded
} else {
bounds := framePair.Media.Bounds()
newDx, newDy := bounds.Dx(), bounds.Dy()
if bs.videoEncoder == nil || dx != newDx || dy != newDy {
dx, dy = newDx, newDy
bs.logger.Infow("detected new image bounds", "width", dx, "height", dy)
if err := bs.initVideoCodec(dx, dy); err != nil {
bs.logger.Error(err)
initErr = true
return
}
}
// thread-safe because the size is static
var err error
encodedFrame, err = bs.videoEncoder.Encode(bs.shutdownCtx, framePair.Media)
if err != nil {
bs.logger.Error(err)
return
}
}
if encodedFrame != nil {
select {
case <-bs.shutdownCtx.Done():
return
case bs.outputVideoChan <- encodedFrame:
}
}
}()
if initErr {
return
}
}
}
func (bs *basicStream) processInputAudioChunks() {
defer close(bs.outputAudioChan)
var samplingRate, channels int
for {
select {
case <-bs.shutdownCtx.Done():
return
default:
}
var audioChunkPair MediaReleasePair[wave.Audio]
select {
case audioChunkPair = <-bs.inputAudioChan:
case <-bs.shutdownCtx.Done():
return
}
if audioChunkPair.Media == nil {
continue
}
var initErr bool
func() {
if audioChunkPair.Release != nil {
defer audioChunkPair.Release()
}
info := audioChunkPair.Media.ChunkInfo()
newSamplingRate, newChannels := info.SamplingRate, info.Channels
if samplingRate != newSamplingRate || channels != newChannels {
samplingRate, channels = newSamplingRate, newChannels
bs.logger.Infow("detected new audio info", "sampling_rate", samplingRate, "channels", channels)
bs.audioTrackLocal.setAudioLatency(bs.audioLatency)
if err := bs.initAudioCodec(samplingRate, channels); err != nil {
bs.logger.Error(err)
initErr = true
return
}
}
encodedChunk, ready, err := bs.audioEncoder.Encode(bs.shutdownCtx, audioChunkPair.Media)
if err != nil {
bs.logger.Error(err)
return
}
if ready && encodedChunk != nil {
select {
case <-bs.shutdownCtx.Done():
return
case bs.outputAudioChan <- encodedChunk:
}
}
}()
if initErr {
return
}
}
}
func (bs *basicStream) processOutputFrames() {
framesSent := 0
for outputFrame := range bs.outputVideoChan {
select {
case <-bs.shutdownCtx.Done():
return
default:
}
now := time.Now()
if err := bs.videoTrackLocal.WriteData(outputFrame); err != nil {
bs.logger.Errorw("error writing frame", "error", err)
}
framesSent++
if Debug {
bs.logger.Debugw("wrote sample", "frames_sent", framesSent, "write_time", time.Since(now))
}
}
}
func (bs *basicStream) processOutputAudioChunks() {
chunksSent := 0
for outputChunk := range bs.outputAudioChan {
select {
case <-bs.shutdownCtx.Done():
return
default:
}
now := time.Now()
if err := bs.audioTrackLocal.WriteData(outputChunk); err != nil {
bs.logger.Errorw("error writing audio chunk", "error", err)
}
chunksSent++
if Debug {
bs.logger.Debugw("wrote sample", "chunks_sent", chunksSent, "write_time", time.Since(now))
}
}
}
func (bs *basicStream) initVideoCodec(width, height int) error {
var err error
bs.videoEncoder, err = bs.config.VideoEncoderFactory.New(width, height, bs.config.TargetFrameRate, bs.logger)
return err
}
func (bs *basicStream) initAudioCodec(sampleRate, channelCount int) error {
var err error
if bs.audioEncoder != nil {
bs.audioEncoder.Close()
}
bs.audioEncoder, err = bs.config.AudioEncoderFactory.New(sampleRate, channelCount, bs.audioLatency, bs.logger)
return err
}